/**
* This file is part of the OpenAL Soft cross platform audio library
*
* Copyright (C) 2019 by Anis A. Hireche
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* * Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
*
* * Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
*
* * Neither the name of Spherical-Harmonic-Transform nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
* AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE
* LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
* INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
* POSSIBILITY OF SUCH DAMAGE.
*/
#include "config.h"
#include <algorithm>
#include <array>
#include <cstdlib>
#include <functional>
#include <iterator>
#include "alc/effects/base.h"
#include "almalloc.h"
#include "alnumbers.h"
#include "alnumeric.h"
#include "alspan.h"
#include "core/ambidefs.h"
#include "core/bufferline.h"
#include "core/context.h"
#include "core/devformat.h"
#include "core/device.h"
#include "core/effectslot.h"
#include "core/mixer.h"
#include "intrusive_ptr.h"
namespace {
using uint = unsigned int;
#define MAX_UPDATE_SAMPLES 256
#define NUM_FORMANTS 4
#define NUM_FILTERS 2
#define Q_FACTOR 5.0f
#define VOWEL_A_INDEX 0
#define VOWEL_B_INDEX 1
#define WAVEFORM_FRACBITS 24
#define WAVEFORM_FRACONE (1<<WAVEFORM_FRACBITS)
#define WAVEFORM_FRACMASK (WAVEFORM_FRACONE-1)
inline float Sin(uint index)
{
constexpr float scale{al::numbers::pi_v<float>*2.0f / WAVEFORM_FRACONE};
return std::sin(static_cast<float>(index) * scale)*0.5f + 0.5f;
}
inline float Saw(uint index)
{ return static_cast<float>(index) / float{WAVEFORM_FRACONE}; }
inline float Triangle(uint index)
{ return std::fabs(static_cast<float>(index)*(2.0f/WAVEFORM_FRACONE) - 1.0f); }
inline float Half(uint) { return 0.5f; }
template<float (&func)(uint)>
void Oscillate(float *RESTRICT dst, uint index, const uint step, size_t todo)
{
for(size_t i{0u};i < todo;i++)
{
index += step;
index &= WAVEFORM_FRACMASK;
dst[i] = func(index);
}
}
struct FormantFilter
{
float mCoeff{0.0f};
float mGain{1.0f};
float mS1{0.0f};
float mS2{0.0f};
FormantFilter() = default;
FormantFilter(float f0norm, float gain)
: mCoeff{std::tan(al::numbers::pi_v<float> * f0norm)}, mGain{gain}
{ }
inline void process(const float *samplesIn, float *samplesOut, const size_t numInput)
{
/* A state variable filter from a topology-preserving transform.
* Based on a talk given by Ivan Cohen: https://www.youtube.com/watch?v=esjHXGPyrhg
*/
const float g{mCoeff};
const float gain{mGain};
const float h{1.0f / (1.0f + (g/Q_FACTOR) + (g*g))};
float s1{mS1};
float s2{mS2};
for(size_t i{0u};i < numInput;i++)
{
const float H{(samplesIn[i] - (1.0f/Q_FACTOR + g)*s1 - s2)*h};
const float B{g*H + s1};
const float L{g*B + s2};
s1 = g*H + B;
s2 = g*B + L;
// Apply peak and accumulate samples.
samplesOut[i] += B * gain;
}
mS1 = s1;
mS2 = s2;
}
inline void clear()
{
mS1 = 0.0f;
mS2 = 0.0f;
}
};
struct VmorpherState final : public EffectState {
struct {
uint mTargetChannel{InvalidChannelIndex};
/* Effect parameters */
FormantFilter mFormants[NUM_FILTERS][NUM_FORMANTS];
/* Effect gains for each channel */
float mCurrentGain{};
float mTargetGain{};
} mChans[MaxAmbiChannels];
void (*mGetSamples)(float*RESTRICT, uint, const uint, size_t){};
uint mIndex{0};
uint mStep{1};
/* Effects buffers */
alignas(16) float mSampleBufferA[MAX_UPDATE_SAMPLES]{};
alignas(16) float mSampleBufferB[MAX_UPDATE_SAMPLES]{};
alignas(16) float mLfo[MAX_UPDATE_SAMPLES]{};
void deviceUpdate(const DeviceBase *device, const BufferStorage *buffer) override;
void update(const ContextBase *context, const EffectSlot *slot, const EffectProps *props,
const EffectTarget target) override;
void process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn,
const al::span<FloatBufferLine> samplesOut) override;
static std::array<FormantFilter,4> getFiltersByPhoneme(VMorpherPhenome phoneme,
float frequency, float pitch);
DEF_NEWDEL(VmorpherState)
};
std::array<FormantFilter,4> VmorpherState::getFiltersByPhoneme(VMorpherPhenome phoneme,
float frequency, float pitch)
{
/* Using soprano formant set of values to
* better match mid-range frequency space.
*
* See: https://www.classes.cs.uchicago.edu/archive/1999/spring/CS295/Computing_Resources/Csound/CsManual3.48b1.HTML/Appendices/table3.html
*/
switch(phoneme)
{
case VMorpherPhenome::A:
return {{
{( 800 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
{(1150 * pitch) / frequency, 0.501187f}, /* std::pow(10.0f, -6 / 20.0f); */
{(2900 * pitch) / frequency, 0.025118f}, /* std::pow(10.0f, -32 / 20.0f); */
{(3900 * pitch) / frequency, 0.100000f} /* std::pow(10.0f, -20 / 20.0f); */
}};
case VMorpherPhenome::E:
return {{
{( 350 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
{(2000 * pitch) / frequency, 0.100000f}, /* std::pow(10.0f, -20 / 20.0f); */
{(2800 * pitch) / frequency, 0.177827f}, /* std::pow(10.0f, -15 / 20.0f); */
{(3600 * pitch) / frequency, 0.009999f} /* std::pow(10.0f, -40 / 20.0f); */
}};
case VMorpherPhenome::I:
return {{
{( 270 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
{(2140 * pitch) / frequency, 0.251188f}, /* std::pow(10.0f, -12 / 20.0f); */
{(2950 * pitch) / frequency, 0.050118f}, /* std::pow(10.0f, -26 / 20.0f); */
{(3900 * pitch) / frequency, 0.050118f} /* std::pow(10.0f, -26 / 20.0f); */
}};
case VMorpherPhenome::O:
return {{
{( 450 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
{( 800 * pitch) / frequency, 0.281838f}, /* std::pow(10.0f, -11 / 20.0f); */
{(2830 * pitch) / frequency, 0.079432f}, /* std::pow(10.0f, -22 / 20.0f); */
{(3800 * pitch) / frequency, 0.079432f} /* std::pow(10.0f, -22 / 20.0f); */
}};
case VMorpherPhenome::U:
return {{
{( 325 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
{( 700 * pitch) / frequency, 0.158489f}, /* std::pow(10.0f, -16 / 20.0f); */
{(2700 * pitch) / frequency, 0.017782f}, /* std::pow(10.0f, -35 / 20.0f); */
{(3800 * pitch) / frequency, 0.009999f} /* std::pow(10.0f, -40 / 20.0f); */
}};
default:
break;
}
return {};
}
void VmorpherState::deviceUpdate(const DeviceBase*, const BufferStorage*)
{
for(auto &e : mChans)
{
e.mTargetChannel = InvalidChannelIndex;
std::for_each(std::begin(e.mFormants[VOWEL_A_INDEX]), std::end(e.mFormants[VOWEL_A_INDEX]),
std::mem_fn(&FormantFilter::clear));
std::for_each(std::begin(e.mFormants[VOWEL_B_INDEX]), std::end(e.mFormants[VOWEL_B_INDEX]),
std::mem_fn(&FormantFilter::clear));
e.mCurrentGain = 0.0f;
}
}
void VmorpherState::update(const ContextBase *context, const EffectSlot *slot,
const EffectProps *props, const EffectTarget target)
{
const DeviceBase *device{context->mDevice};
const float frequency{static_cast<float>(device->Frequency)};
const float step{props->Vmorpher.Rate / frequency};
mStep = fastf2u(clampf(step*WAVEFORM_FRACONE, 0.0f, float{WAVEFORM_FRACONE-1}));
if(mStep == 0)
mGetSamples = Oscillate<Half>;
else if(props->Vmorpher.Waveform == VMorpherWaveform::Sinusoid)
mGetSamples = Oscillate<Sin>;
else if(props->Vmorpher.Waveform == VMorpherWaveform::Triangle)
mGetSamples = Oscillate<Triangle>;
else /*if(props->Vmorpher.Waveform == VMorpherWaveform::Sawtooth)*/
mGetSamples = Oscillate<Saw>;
const float pitchA{std::pow(2.0f,
static_cast<float>(props->Vmorpher.PhonemeACoarseTuning) / 12.0f)};
const float pitchB{std::pow(2.0f,
static_cast<float>(props->Vmorpher.PhonemeBCoarseTuning) / 12.0f)};
auto vowelA = getFiltersByPhoneme(props->Vmorpher.PhonemeA, frequency, pitchA);
auto vowelB = getFiltersByPhoneme(props->Vmorpher.PhonemeB, frequency, pitchB);
/* Copy the filter coefficients to the input channels. */
for(size_t i{0u};i < slot->Wet.Buffer.size();++i)
{
std::copy(vowelA.begin(), vowelA.end(), std::begin(mChans[i].mFormants[VOWEL_A_INDEX]));
std::copy(vowelB.begin(), vowelB.end(), std::begin(mChans[i].mFormants[VOWEL_B_INDEX]));
}
mOutTarget = target.Main->Buffer;
auto set_channel = [this](size_t idx, uint outchan, float outgain)
{
mChans[idx].mTargetChannel = outchan;
mChans[idx].mTargetGain = outgain;
};
target.Main->setAmbiMixParams(slot->Wet, slot->Gain, set_channel);
}
void VmorpherState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
{
/* Following the EFX specification for a conformant implementation which describes
* the effect as a pair of 4-band formant filters blended together using an LFO.
*/
for(size_t base{0u};base < samplesToDo;)
{
const size_t td{minz(MAX_UPDATE_SAMPLES, samplesToDo-base)};
mGetSamples(mLfo, mIndex, mStep, td);
mIndex += static_cast<uint>(mStep * td);
mIndex &= WAVEFORM_FRACMASK;
auto chandata = std::begin(mChans);
for(const auto &input : samplesIn)
{
const size_t outidx{chandata->mTargetChannel};
if(outidx == InvalidChannelIndex)
{
++chandata;
continue;
}
auto& vowelA = chandata->mFormants[VOWEL_A_INDEX];
auto& vowelB = chandata->mFormants[VOWEL_B_INDEX];
/* Process first vowel. */
std::fill_n(std::begin(mSampleBufferA), td, 0.0f);
vowelA[0].process(&input[base], mSampleBufferA, td);
vowelA[1].process(&input[base], mSampleBufferA, td);
vowelA[2].process(&input[base], mSampleBufferA, td);
vowelA[3].process(&input[base], mSampleBufferA, td);
/* Process second vowel. */
std::fill_n(std::begin(mSampleBufferB), td, 0.0f);
vowelB[0].process(&input[base], mSampleBufferB, td);
vowelB[1].process(&input[base], mSampleBufferB, td);
vowelB[2].process(&input[base], mSampleBufferB, td);
vowelB[3].process(&input[base], mSampleBufferB, td);
alignas(16) float blended[MAX_UPDATE_SAMPLES];
for(size_t i{0u};i < td;i++)
blended[i] = lerpf(mSampleBufferA[i], mSampleBufferB[i], mLfo[i]);
/* Now, mix the processed sound data to the output. */
MixSamples({blended, td}, samplesOut[outidx].data()+base, chandata->mCurrentGain,
chandata->mTargetGain, samplesToDo-base);
++chandata;
}
base += td;
}
}
struct VmorpherStateFactory final : public EffectStateFactory {
al::intrusive_ptr<EffectState> create() override
{ return al::intrusive_ptr<EffectState>{new VmorpherState{}}; }
};
} // namespace
EffectStateFactory *VmorpherStateFactory_getFactory()
{
static VmorpherStateFactory VmorpherFactory{};
return &VmorpherFactory;
}